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...other will be an outbound call via a second T1 interface. Does anyone have a working configuration for an Asterisk initiated 3 way call? Thanks and Regards Shad Mortazavi ------------------------------------------------------ Nexus Group Technical Manager n|m Nexus Management Inc SIP: 0401@pbx.nexusmgmt.com ?

...k 1.2.4. Ser is listening on 5060 and asterisk on 5065. The setup is that people use serweb to create an account and register a phone. Their calls are routed from ser to asterisk and then inbound on IAX2. The server has a public and an internal interface. The real FQDN of the server is nmibwksip3.nexusmgmt.com and it has cnames of pbx and nexphone. The pbx name is used to route calls to the asterisk. Setting up inbound calls works with this routing statement in ser.cfg: if (uri =~ "sip:[0-9]{4}@pbx\.nexusmgmt\.com") { forward(65.126.236.148,5065); break; }; The problem is that the call...

Dear Group, I have just purchased two Linksys SPA941 and flashed these to the latest firmware. Everything works well except for the Hold button? Has anyone else experienced the same issue? What was the solution? Kind Regards Shad Mortazavi

Dear Group, Does any one have experience using SoundPoint(r) IP 300? I have one call center on Snom 200's I'm adding a second and was looking at the SoundPoint, but needed some input. Thanks Shad Mortazavi --------------------------------------------------- Nexus Technical Manager n|m Nexus Management Inc Sydney -------------- next part -------------- An HTML attachment was

Dear All, I'm running version 0.7.1 of Asterisk server for our global help desk. We have put together a comprehensive reporting package for static's from the CDR. I'm not able to calculate the time a call is in the queue before it goes to an agent? I would appreciate help with working this out. Warm Regards and Thanks Shad Mortazavi

...pass, we receive errors about "could not grab keyboard" and hints that there might be malice afoot. What is causing this error and how to we go about eliminating it? Thanks - John Sullivan -- John A. Sullivan III Chief Technology Officer Nexus Management +1 207-985-7880 john.sullivan at nexusmgmt.com --- If you are interested in helping to develop a GPL enterprise class VPN/Firewall/Security device management console, please visit http://iscs.sourceforge.net

Does anyone have any feelings one way or the other about running Asterisk on AMD vs running Asterisk on Intel? Thanks, Lyle J. McKarns ------------------------------------------- Networking/Linux Engineering Team n|m Nexus Management 4 Industrial Parkway Suite 101 Brunswick, Maine 04011 Tel (USA) : 1 207 319 1105 Tel (UK) : 0207 100 4968 Fax : 1 207 725 8552 Nexus Management,

Dear Group, I'm in the final phases of switching over from my existing PBX to an Asterisk based PBX. On my current PBX calls are routed on the existing PBX using a assigned DNSI number, and I'm looking at replicating this functionality. Does anyone have experience in routing calls from a T1 based on a DNSI number? If so would you mind; a) Confirming this functionality and b) giving

Dear All, I have a number of call queues defined in Asterisk. I would like to program a system attendant that tells people; 1. Every 60 seconds 'Your call will be answered as soon as possible' 2. Tell the user how many calls are on the queue. I would then like them put back on hold music. Does someone have a configuration for this or something similar? Your help would be greatly

Dear all, I have spent some time tying to get the system attendant patch to work; http://bugs.digium.com/bug_view_page.php?bug_id=0000214 I get no errors patching the system and the function runs, but I keep getting the following error; queue: Nexus1, options: (null), url: (null), announce: (null), timeout: 0 -- Started music on hold, class 'default', on SIP/phone10-a3f0 --

Dear All, I have move from the USA to Sydney, Australia. I have gone from a data center environment at work and cable at home to a 513k/128k ADSL line. I'm experiencing two issues; 1) There is a latency of .5 - .8 seconds between me and the USA. 2) I have been in two calls where my voice has been describes as 'Scratchy'? I'm using a SIP Phone from SJ Phone, and a Plantronics

Dear Group, At the moment I use SJPhone as my soft phone with Asterisk. I prefer the look and feel of the x-ten lite. However, when ever I use my x-ten lite I get a lot of breakup in my communication. E.g. I will play some hold music, and every 5-6 seconds I drop some packets. I don't have the same issue with SJPhone. I'm sure this is a configuration issues, but I can work

Dear All, I'm looking for a presence utility for Asterisk, either to work as a standalone application or integrated into a platform like Jabber. Any ideas? Warm Regards Shad Mortazavi --------------------------------------------------- Nexus Technical Manager n|m Nexus Management Inc Sydney -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear All, I'm running asterisk 0.7.1 on a production call center and second call center on Asterisk CVS-HEAD-05/18/04-01:57:42. On the Asterisk CVS-HEAD-05/18/04-01:57:42 box I have a queue_log file that I can use to get statistics. I get no such file in 0.7.1. Is this release dependent or is there configuration file I'm missing. Thanks and Regards Shad Mortazavi

Dear All, I would like to add this feature to my version of asterisk. Is there a patch I can apply to get this function? Does anyone have any instructions for this? Thanks Shad Mortazavi ------------------------------------------------------ Nexus Global Technical Manager n|m Nexus Management Inc

Dear Group, I have my Asterisk box working with a Mediatrix 1204. I have 2 questions; 1) I do not seem to get a Call ID on the call coming via the Mediatrix 1204. I was wondering if anyone had this configured and if they could share this with me? 2) How do you route a call based on caller ID on Asterisk. At the moment I'm routing calls via DNIS. Thanks and Regards Shad Mortazavi

Dear Group, I have been able to configure my Asterisk BOX to receive calls from Mediatrix 1204. I'm having problems sending calls out via my Mediatrix unit. The SIP Invite is sent to the Mediatrix but the Mediatrix unit sends back a Status : 480 Temporarily Unavailable. This is my configuration on Asterisk; exten => _78996.,1,Dial(SIP/${EXTEN:5}@192.168.6.52) exten =>

Dear All, I have the following setup. Quad T1's<->Asterisk (PBX)<->(LAN<->DMZ)<->SER<->(Firewall)<->(Internet) | Local US Help Desk (Snom 200') This setup works well. I can pass calls from over the internet to the Asterisk PBX via SER using X-Ten Lit. I have a couple of questions; 1. How do I

Dear Forum, I'm using the AgentCallbackLogin function to log my agents onto multiple call queues. exten => 3001,1, AgenCallbackLogin(1001,@sip). This works very well. I can not work out how to log them back out? On of the forum members was kind enough to point me into the directions of 'dial a null extension and press * to logout'. I don't seem to be able to translate

Dear Group, I need to write a couple of reporting tools for my Call Center Asterisks implementation. I have multiple call queues with multiple agents that can sign in and based on gain access to multiple queues based on their assignments. I would like to write a script to collect call statistics for the agents the queues and the calls, and to put these into MySQL for reporting purposes.